Optional: Use nullopt and implicit construction in /modules/audio_coding
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=kwiberg@webrtc.org
Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 360a583..8f8b273 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -103,7 +103,7 @@
last_audio_decoder_ = ci;
last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
RTC_DCHECK(last_audio_format_);
- last_packet_sample_rate_hz_ = rtc::Optional<int>(ci->plfreq);
+ last_packet_sample_rate_hz_ = ci->plfreq;
}
} // |crit_sect_| is released.
@@ -272,9 +272,9 @@
void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
neteq_->RemoveAllPayloadTypes();
- last_audio_decoder_ = rtc::Optional<CodecInst>();
- last_audio_format_ = rtc::Optional<SdpAudioFormat>();
- last_packet_sample_rate_hz_ = rtc::Optional<int>();
+ last_audio_decoder_ = rtc::nullopt;
+ last_audio_format_ = rtc::nullopt;
+ last_packet_sample_rate_hz_ = rtc::nullopt;
}
int AcmReceiver::RemoveCodec(uint8_t payload_type) {
@@ -285,9 +285,9 @@
return -1;
}
if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
- last_audio_decoder_ = rtc::Optional<CodecInst>();
- last_audio_format_ = rtc::Optional<SdpAudioFormat>();
- last_packet_sample_rate_hz_ = rtc::Optional<int>();
+ last_audio_decoder_ = rtc::nullopt;
+ last_audio_format_ = rtc::nullopt;
+ last_packet_sample_rate_hz_ = rtc::nullopt;
}
return 0;
}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index c420c7e..e8f8b4a 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -483,8 +483,7 @@
// of type "speech."
ASSERT_TRUE(packet_sent_);
ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
- EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
- receiver_->last_packet_sample_rate_hz());
+ EXPECT_EQ(c.inst.plfreq, receiver_->last_packet_sample_rate_hz());
// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
// the expected codec.
@@ -496,8 +495,7 @@
InsertOnePacketOfSilence(c.id);
ASSERT_TRUE(packet_sent_);
}
- EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
- receiver_->last_packet_sample_rate_hz());
+ EXPECT_EQ(c.inst.plfreq, receiver_->last_packet_sample_rate_hz());
EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
EXPECT_TRUE(CodecsEqual(c.inst, codec));
}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 361278f..340cc92 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -606,21 +606,20 @@
if (encoder_factory_) {
auto* ci = encoder_factory_->codec_manager.GetCodecInst();
if (ci) {
- return rtc::Optional<CodecInst>(*ci);
+ return *ci;
}
CreateSpeechEncoderIfNecessary(encoder_factory_.get());
const std::unique_ptr<AudioEncoder>& enc =
encoder_factory_->codec_manager.GetStackParams()->speech_encoder;
if (enc) {
- return rtc::Optional<CodecInst>(
- acm2::CodecManager::ForgeCodecInst(enc.get()));
+ return acm2::CodecManager::ForgeCodecInst(enc.get());
}
- return rtc::Optional<CodecInst>();
+ return rtc::nullopt;
} else {
return encoder_stack_
? rtc::Optional<CodecInst>(
acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
- : rtc::Optional<CodecInst>();
+ : rtc::nullopt;
}
}
@@ -639,8 +638,7 @@
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_) {
- encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps,
- rtc::Optional<int64_t>());
+ encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, rtc::nullopt);
}
}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index ca59b31..68dd671 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -184,8 +184,7 @@
// Set up L16 codec.
virtual void SetUpL16Codec() {
- audio_format_ =
- rtc::Optional<SdpAudioFormat>(SdpAudioFormat("L16", kSampleRateHz, 1));
+ audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1));
codec_.pltype = kPayloadType;
}
@@ -660,8 +659,7 @@
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
- audio_format_ =
- rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", kSampleRateHz, 1));
+ audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1);
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
diff --git a/modules/audio_coding/acm2/codec_manager.cc b/modules/audio_coding/acm2/codec_manager.cc
index 50ef9ef..a101d3d 100644
--- a/modules/audio_coding/acm2/codec_manager.cc
+++ b/modules/audio_coding/acm2/codec_manager.cc
@@ -106,7 +106,7 @@
codec_stack_params_.use_cng = false;
}
- send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
+ send_codec_inst_ = send_codec;
recreate_encoder_ = true; // Caller must recreate it.
return true;
}
diff --git a/modules/audio_coding/acm2/codec_manager.h b/modules/audio_coding/acm2/codec_manager.h
index a2a6810..7485426 100644
--- a/modules/audio_coding/acm2/codec_manager.h
+++ b/modules/audio_coding/acm2/codec_manager.h
@@ -43,7 +43,7 @@
return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
}
- void UnsetCodecInst() { send_codec_inst_ = rtc::Optional<CodecInst>(); }
+ void UnsetCodecInst() { send_codec_inst_ = rtc::nullopt; }
const RentACodec::StackParameters* GetStackParams() const {
return &codec_stack_params_;
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index 39efd96..78db38d 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -55,7 +55,7 @@
rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
rtc::Optional<int> mi = CodecIndexFromId(codec_id);
return mi ? rtc::Optional<CodecInst>(Database()[*mi])
- : rtc::Optional<CodecInst>();
+ : rtc::nullopt;
}
rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
@@ -69,7 +69,7 @@
rtc::Optional<CodecId> codec_id =
CodecIdByParams(payload_name, sampling_freq_hz, channels);
if (!codec_id)
- return rtc::Optional<CodecInst>();
+ return rtc::nullopt;
rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
RTC_DCHECK(ci);
@@ -90,7 +90,7 @@
return i ? rtc::Optional<bool>(
ACMCodecDB::codec_settings_[*i].channel_support >=
num_channels)
- : rtc::Optional<bool>();
+ : rtc::nullopt;
}
rtc::ArrayView<const CodecInst> RentACodec::Database() {
@@ -103,12 +103,11 @@
size_t num_channels) {
rtc::Optional<int> i = CodecIndexFromId(codec_id);
if (!i)
- return rtc::Optional<NetEqDecoder>();
+ return rtc::nullopt;
const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
- return rtc::Optional<NetEqDecoder>(
- (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
- ? NetEqDecoder::kDecoderOpus_2ch
- : ned);
+ return (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
+ ? NetEqDecoder::kDecoderOpus_2ch
+ : ned;
}
RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
@@ -277,7 +276,7 @@
auto pt = [¶m](const std::map<int, int>& m) {
auto it = m.find(param->speech_encoder->SampleRateHz());
- return it == m.end() ? rtc::Optional<int>()
+ return it == m.end() ? rtc::nullopt
: rtc::Optional<int>(it->second);
};
auto cng_pt = pt(param->cng_payload_types);
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
index cecb914..f8fac4c 100644
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ b/modules/audio_coding/acm2/rent_a_codec.h
@@ -111,14 +111,14 @@
const int i = static_cast<int>(codec_id);
return i >= 0 && i < static_cast<int>(NumberOfCodecs())
? rtc::Optional<int>(i)
- : rtc::Optional<int>();
+ : rtc::nullopt;
}
static inline rtc::Optional<CodecId> CodecIdFromIndex(int codec_index) {
return static_cast<size_t>(codec_index) < NumberOfCodecs()
? rtc::Optional<RentACodec::CodecId>(
static_cast<RentACodec::CodecId>(codec_index))
- : rtc::Optional<RentACodec::CodecId>();
+ : rtc::nullopt;
}
static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 54423e6..55e5309 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -58,68 +58,61 @@
AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default;
void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
- last_metrics_.uplink_bandwidth_bps = rtc::Optional<int>(uplink_bandwidth_bps);
+ last_metrics_.uplink_bandwidth_bps = uplink_bandwidth_bps;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
- network_metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(uplink_bandwidth_bps);
+ network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
- last_metrics_.uplink_packet_loss_fraction =
- rtc::Optional<float>(uplink_packet_loss_fraction);
+ last_metrics_.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
- network_metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(uplink_packet_loss_fraction);
+ network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetUplinkRecoverablePacketLossFraction(
float uplink_recoverable_packet_loss_fraction) {
last_metrics_.uplink_recoverable_packet_loss_fraction =
- rtc::Optional<float>(uplink_recoverable_packet_loss_fraction);
+ uplink_recoverable_packet_loss_fraction;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.uplink_recoverable_packet_loss_fraction =
- rtc::Optional<float>(uplink_recoverable_packet_loss_fraction);
+ uplink_recoverable_packet_loss_fraction;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
- last_metrics_.rtt_ms = rtc::Optional<int>(rtt_ms);
+ last_metrics_.rtt_ms = rtt_ms;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
- network_metrics.rtt_ms = rtc::Optional<int>(rtt_ms);
+ network_metrics.rtt_ms = rtt_ms;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetTargetAudioBitrate(
int target_audio_bitrate_bps) {
- last_metrics_.target_audio_bitrate_bps =
- rtc::Optional<int>(target_audio_bitrate_bps);
+ last_metrics_.target_audio_bitrate_bps = target_audio_bitrate_bps;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
- network_metrics.target_audio_bitrate_bps =
- rtc::Optional<int>(target_audio_bitrate_bps);
+ network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetOverhead(size_t overhead_bytes_per_packet) {
- last_metrics_.overhead_bytes_per_packet =
- rtc::Optional<size_t>(overhead_bytes_per_packet);
+ last_metrics_.overhead_bytes_per_packet = overhead_bytes_per_packet;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
- network_metrics.overhead_bytes_per_packet =
- rtc::Optional<size_t>(overhead_bytes_per_packet);
+ network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
UpdateNetworkMetrics(network_metrics);
}
@@ -131,7 +124,7 @@
// Update ANA stats.
auto increment_opt = [](rtc::Optional<uint32_t>& a) {
- a = rtc::Optional<uint32_t>(a.value_or(0) + 1);
+ a = a.value_or(0) + 1;
};
if (prev_config_) {
if (config.bitrate_bps != prev_config_->bitrate_bps) {
@@ -154,11 +147,10 @@
increment_opt(stats_.channel_action_counter);
}
if (config.uplink_packet_loss_fraction) {
- stats_.uplink_packet_loss_fraction =
- rtc::Optional<float>(*config.uplink_packet_loss_fraction);
+ stats_.uplink_packet_loss_fraction = *config.uplink_packet_loss_fraction;
}
}
- prev_config_ = rtc::Optional<AudioEncoderRuntimeConfig>(config);
+ prev_config_ = config;
// Prevent certain controllers from taking action (determined by field trials)
if (!enable_bitrate_adaptation_ && config.bitrate_bps) {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 1f0f71a..c437918 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -124,7 +124,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr int kBandwidth = 16000;
Controller::NetworkMetrics check;
- check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
+ check.uplink_bandwidth_bps = kBandwidth;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
}
@@ -134,7 +134,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr float kPacketLoss = 0.7f;
Controller::NetworkMetrics check;
- check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
+ check.uplink_packet_loss_fraction = kPacketLoss;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
}
@@ -144,8 +144,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr float kRecoverablePacketLoss = 0.1f;
Controller::NetworkMetrics check;
- check.uplink_recoverable_packet_loss_fraction =
- rtc::Optional<float>(kRecoverablePacketLoss);
+ check.uplink_recoverable_packet_loss_fraction = kRecoverablePacketLoss;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
kRecoverablePacketLoss);
@@ -155,7 +154,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr int kRtt = 100;
Controller::NetworkMetrics check;
- check.rtt_ms = rtc::Optional<int>(kRtt);
+ check.rtt_ms = kRtt;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetRtt(kRtt);
}
@@ -165,7 +164,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr int kTargetAudioBitrate = 15000;
Controller::NetworkMetrics check;
- check.target_audio_bitrate_bps = rtc::Optional<int>(kTargetAudioBitrate);
+ check.target_audio_bitrate_bps = kTargetAudioBitrate;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
}
@@ -174,7 +173,7 @@
auto states = CreateAudioNetworkAdaptor();
constexpr size_t kOverhead = 64;
Controller::NetworkMetrics check;
- check.overhead_bytes_per_packet = rtc::Optional<size_t>(kOverhead);
+ check.overhead_bytes_per_packet = kOverhead;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetOverhead(kOverhead);
}
@@ -196,8 +195,8 @@
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
auto states = CreateAudioNetworkAdaptor();
AudioEncoderRuntimeConfig config;
- config.bitrate_bps = rtc::Optional<int>(32000);
- config.enable_fec = rtc::Optional<bool>(true);
+ config.bitrate_bps = 32000;
+ config.enable_fec = true;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config));
@@ -223,7 +222,7 @@
constexpr size_t kOverhead = 64;
Controller::NetworkMetrics check;
- check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
+ check.uplink_bandwidth_bps = kBandwidth;
int64_t timestamp_check = kClockInitialTimeMs;
EXPECT_CALL(*states.mock_debug_dump_writer,
@@ -232,15 +231,14 @@
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100));
timestamp_check += 100;
- check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
+ check.uplink_packet_loss_fraction = kPacketLoss;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(50));
timestamp_check += 50;
- check.uplink_recoverable_packet_loss_fraction =
- rtc::Optional<float>(kRecoverablePacketLoss);
+ check.uplink_recoverable_packet_loss_fraction = kRecoverablePacketLoss;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
@@ -248,21 +246,21 @@
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(200));
timestamp_check += 200;
- check.rtt_ms = rtc::Optional<int>(kRtt);
+ check.rtt_ms = kRtt;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetRtt(kRtt);
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(150));
timestamp_check += 150;
- check.target_audio_bitrate_bps = rtc::Optional<int>(kTargetAudioBitrate);
+ check.target_audio_bitrate_bps = kTargetAudioBitrate;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(50));
timestamp_check += 50;
- check.overhead_bytes_per_packet = rtc::Optional<size_t>(kOverhead);
+ check.overhead_bytes_per_packet = kOverhead;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetOverhead(kOverhead);
@@ -275,8 +273,8 @@
auto states = CreateAudioNetworkAdaptor();
AudioEncoderRuntimeConfig config;
- config.bitrate_bps = rtc::Optional<int>(32000);
- config.enable_fec = rtc::Optional<bool>(true);
+ config.bitrate_bps = 32000;
+ config.enable_fec = true;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config));
@@ -291,18 +289,18 @@
// Simulate some adaptation, otherwise the stats will not show anything.
AudioEncoderRuntimeConfig config1, config2;
- config1.bitrate_bps = rtc::Optional<int>(32000);
- config1.num_channels = rtc::Optional<size_t>(2);
- config1.enable_fec = rtc::Optional<bool>(true);
- config1.enable_dtx = rtc::Optional<bool>(true);
- config1.frame_length_ms = rtc::Optional<int>(120);
- config1.uplink_packet_loss_fraction = rtc::Optional<float>(0.1f);
- config2.bitrate_bps = rtc::Optional<int>(16000);
- config2.num_channels = rtc::Optional<size_t>(1);
- config2.enable_fec = rtc::Optional<bool>(false);
- config2.enable_dtx = rtc::Optional<bool>(false);
- config2.frame_length_ms = rtc::Optional<int>(60);
- config1.uplink_packet_loss_fraction = rtc::Optional<float>(0.1f);
+ config1.bitrate_bps = 32000;
+ config1.num_channels = 2;
+ config1.enable_fec = true;
+ config1.enable_dtx = true;
+ config1.frame_length_ms = 120;
+ config1.uplink_packet_loss_fraction = 0.1f;
+ config2.bitrate_bps = 16000;
+ config2.num_channels = 1;
+ config2.enable_fec = false;
+ config2.enable_dtx = false;
+ config2.frame_length_ms = 60;
+ config1.uplink_packet_loss_fraction = 0.1f;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config1));
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index 16c4bc4..6850926 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -69,7 +69,7 @@
(*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_);
bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
}
- config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
+ config->bitrate_bps = bitrate_bps_;
}
} // namespace audio_network_adaptor
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index 09d1066..daad293 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -43,7 +43,7 @@
AudioEncoderRuntimeConfig config;
config.frame_length_ms = frame_length_ms;
controller->MakeDecision(&config);
- EXPECT_EQ(rtc::Optional<int>(expected_bitrate_bps), config.bitrate_bps);
+ EXPECT_EQ(expected_bitrate_bps, config.bitrate_bps);
}
} // namespace
@@ -58,10 +58,8 @@
constexpr size_t kOverheadBytesPerPacket = 64;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(kInitialFrameLengthMs * 2),
- kInitialBitrateBps);
+ UpdateNetworkMetrics(&controller, rtc::nullopt, kOverheadBytesPerPacket);
+ CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
TEST(AnaBitrateControllerTest, OutputInitValueWhenOverheadUnknown) {
@@ -70,10 +68,8 @@
constexpr int kTargetBitrateBps = 48000;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>());
- CheckDecision(&controller, rtc::Optional<int>(kInitialFrameLengthMs * 2),
- kInitialBitrateBps);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, rtc::nullopt);
+ CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
@@ -89,10 +85,8 @@
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
// Frame length unchanged, bitrate changes in accordance with
// |metrics.target_audio_bitrate_bps| and |metrics.overhead_bytes_per_packet|.
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(kInitialFrameLengthMs),
- kBitrateBps);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
}
TEST(AnaBitrateControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
@@ -114,13 +108,10 @@
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
Controller::NetworkMetrics network_metrics;
- network_metrics.target_audio_bitrate_bps =
- rtc::Optional<int>(kTargetBitrateBps);
- network_metrics.overhead_bytes_per_packet =
- rtc::Optional<size_t>(kOverheadBytesPerPacket);
+ network_metrics.target_audio_bitrate_bps = kTargetBitrateBps;
+ network_metrics.overhead_bytes_per_packet = kOverheadBytesPerPacket;
controller.UpdateNetworkMetrics(network_metrics);
- CheckDecision(&controller, rtc::Optional<int>(kInitialFrameLengthMs),
- kBitrateBps);
+ CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
}
TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
@@ -134,9 +125,8 @@
constexpr int kBitrateBps =
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(), kBitrateBps);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, rtc::nullopt, kBitrateBps);
}
TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
@@ -151,17 +141,15 @@
constexpr int kBitrateBps =
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(), kBitrateBps);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, rtc::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 60;
constexpr size_t kPacketOverheadRateDiff =
kOverheadBytesPerPacket * 8 * 1000 / 20 -
kOverheadBytesPerPacket * 8 * 1000 / 60;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(kFrameLengthMs),
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, kFrameLengthMs,
kBitrateBps + kPacketOverheadRateDiff);
}
@@ -177,17 +165,15 @@
constexpr int kBitrateBps =
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(), kBitrateBps);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, rtc::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 20;
constexpr size_t kPacketOverheadRateDiff =
kOverheadBytesPerPacket * 8 * 1000 / 20 -
kOverheadBytesPerPacket * 8 * 1000 / 60;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(kFrameLengthMs),
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, kFrameLengthMs,
kBitrateBps - kPacketOverheadRateDiff);
}
@@ -200,9 +186,8 @@
// Set a target rate smaller than overhead rate, the bitrate is bounded by 0.
constexpr int kTargetBitrateBps =
kOverheadBytesPerPacket * 8 * 1000 / kFrameLengthMs - 1;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
- CheckDecision(&controller, rtc::Optional<int>(kFrameLengthMs), 0);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+ CheckDecision(&controller, kFrameLengthMs, 0);
}
TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
@@ -217,46 +202,36 @@
int current_bitrate = rtc::checked_cast<int>(
overall_bitrate - overhead_bytes_per_packet * 8 * 1000 / frame_length_ms);
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
// Next: increase overall bitrate.
overall_bitrate += 100;
current_bitrate += 100;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
// Next: change frame length.
frame_length_ms = 60;
current_bitrate += rtc::checked_cast<int>(
overhead_bytes_per_packet * 8 * 1000 / 20 -
overhead_bytes_per_packet * 8 * 1000 / 60);
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
// Next: change overhead.
overhead_bytes_per_packet -= 30;
current_bitrate += 30 * 8 * 1000 / frame_length_ms;
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
// Next: change frame length.
frame_length_ms = 20;
current_bitrate -= rtc::checked_cast<int>(
overhead_bytes_per_packet * 8 * 1000 / 20 -
overhead_bytes_per_packet * 8 * 1000 / 60);
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
// Next: decrease overall bitrate and frame length.
overall_bitrate -= 100;
@@ -266,10 +241,8 @@
overhead_bytes_per_packet * 8 * 1000 / 20 -
overhead_bytes_per_packet * 8 * 1000 / 60);
- UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
- rtc::Optional<size_t>(overhead_bytes_per_packet));
- CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
- current_bitrate);
+ UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+ CheckDecision(&controller, frame_length_ms, current_bitrate);
}
} // namespace audio_network_adaptor
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
index 55a913a..a1c30db 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -55,7 +55,7 @@
std::min(static_cast<size_t>(2), config_.num_encoder_channels);
}
}
- config->num_channels = rtc::Optional<size_t>(channels_to_encode_);
+ config->num_channels = channels_to_encode_;
}
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index 73160fe..64e5dae 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -41,7 +41,7 @@
}
AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
- EXPECT_EQ(rtc::Optional<size_t>(expected_num_channels), config.num_channels);
+ EXPECT_EQ(expected_num_channels, config.num_channels);
}
} // namespace
@@ -49,37 +49,35 @@
TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr int kInitChannels = 2;
auto controller = CreateChannelController(kInitChannels);
- CheckDecision(controller.get(), rtc::Optional<int>(), kInitChannels);
+ CheckDecision(controller.get(), rtc::nullopt, kInitChannels);
}
TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
constexpr int kInitChannels = 1;
auto controller = CreateChannelController(kInitChannels);
// Use high bandwidth to check output switch to 2.
- CheckDecision(controller.get(), rtc::Optional<int>(kChannel1To2BandwidthBps),
- 2);
+ CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
}
TEST(ChannelControllerTest, SwitchTo1ChannelOnLowUplinkBandwidth) {
constexpr int kInitChannels = 2;
auto controller = CreateChannelController(kInitChannels);
// Use low bandwidth to check output switch to 1.
- CheckDecision(controller.get(), rtc::Optional<int>(kChannel2To1BandwidthBps),
- 1);
+ CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
}
TEST(ChannelControllerTest, Maintain1ChannelOnMediumUplinkBandwidth) {
constexpr int kInitChannels = 1;
auto controller = CreateChannelController(kInitChannels);
// Use between-thresholds bandwidth to check output remains at 1.
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps), 1);
+ CheckDecision(controller.get(), kMediumBandwidthBps, 1);
}
TEST(ChannelControllerTest, Maintain2ChannelsOnMediumUplinkBandwidth) {
constexpr int kInitChannels = 2;
auto controller = CreateChannelController(kInitChannels);
// Use between-thresholds bandwidth to check output remains at 2.
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps), 2);
+ CheckDecision(controller.get(), kMediumBandwidthBps, 2);
}
TEST(ChannelControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
@@ -87,18 +85,16 @@
auto controller = CreateChannelController(kInitChannels);
// Use between-thresholds bandwidth to check output remains at 1.
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps), 1);
+ CheckDecision(controller.get(), kMediumBandwidthBps, 1);
// Use high bandwidth to check output switch to 2.
- CheckDecision(controller.get(), rtc::Optional<int>(kChannel1To2BandwidthBps),
- 2);
+ CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
// Use between-thresholds bandwidth to check output remains at 2.
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps), 2);
+ CheckDecision(controller.get(), kMediumBandwidthBps, 2);
// Use low bandwidth to check output switch to 1.
- CheckDecision(controller.get(), rtc::Optional<int>(kChannel2To1BandwidthBps),
- 1);
+ CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
}
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index d573960..313aa62 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -327,7 +327,7 @@
const std::map<const Controller*, std::pair<int, float>>& scoring_points)
: config_(config),
controllers_(std::move(controllers)),
- last_reordering_time_ms_(rtc::Optional<int64_t>()),
+ last_reordering_time_ms_(rtc::nullopt),
last_scoring_point_(0, 0.0) {
for (auto& controller : controllers_)
default_sorted_controllers_.push_back(controller.get());
@@ -389,7 +389,7 @@
if (sorted_controllers_ != sorted_controllers) {
sorted_controllers_ = sorted_controllers;
- last_reordering_time_ms_ = rtc::Optional<int64_t>(now_ms);
+ last_reordering_time_ms_ = now_ms;
last_scoring_point_ = scoring_point;
}
return sorted_controllers_;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index 846397a..576661c 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -124,41 +124,38 @@
auto states = CreateControllerManager();
// |network_metrics| are empty, and the controllers are supposed to follow the
// default order.
- CheckControllersOrder(&states, rtc::Optional<int>(), rtc::Optional<float>(),
+ CheckControllersOrder(&states, rtc::nullopt, rtc::nullopt,
{0, 1, 2, 3});
}
TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
auto states = CreateControllerManager();
- CheckControllersOrder(&states, rtc::Optional<int>(0),
- rtc::Optional<float>(0.0),
+ CheckControllersOrder(&states, 0,
+ 0.0,
{kNumControllers - 2, kNumControllers - 1, -1, -1});
}
TEST(ControllerManagerTest, ControllersWithCharPointDependOnNetworkMetrics) {
auto states = CreateControllerManager();
- CheckControllersOrder(
- &states, rtc::Optional<int>(kChracteristicBandwithBps[1]),
- rtc::Optional<float>(kChracteristicPacketLossFraction[1]),
- {kNumControllers - 2, kNumControllers - 1, 1, 0});
+ CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+ kChracteristicPacketLossFraction[1],
+ {kNumControllers - 2, kNumControllers - 1, 1, 0});
}
TEST(ControllerManagerTest, DoNotReorderBeforeMinReordingTime) {
rtc::ScopedFakeClock fake_clock;
auto states = CreateControllerManager();
- CheckControllersOrder(
- &states, rtc::Optional<int>(kChracteristicBandwithBps[0]),
- rtc::Optional<float>(kChracteristicPacketLossFraction[0]),
- {kNumControllers - 2, kNumControllers - 1, 0, 1});
+ CheckControllersOrder(&states, kChracteristicBandwithBps[0],
+ kChracteristicPacketLossFraction[0],
+ {kNumControllers - 2, kNumControllers - 1, 0, 1});
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs - 1));
// Move uplink bandwidth and packet loss fraction to the other controller's
// characteristic point, which would cause controller manager to reorder the
// controllers if time had reached min reordering time.
- CheckControllersOrder(
- &states, rtc::Optional<int>(kChracteristicBandwithBps[1]),
- rtc::Optional<float>(kChracteristicPacketLossFraction[1]),
- {kNumControllers - 2, kNumControllers - 1, 0, 1});
+ CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+ kChracteristicPacketLossFraction[1],
+ {kNumControllers - 2, kNumControllers - 1, 0, 1});
}
TEST(ControllerManagerTest, ReorderBeyondMinReordingTimeAndMinDistance) {
@@ -171,16 +168,14 @@
2.0f;
// Set network metrics to be in the middle between the characteristic points
// of two controllers.
- CheckControllersOrder(&states, rtc::Optional<int>(kBandwidthBps),
- rtc::Optional<float>(kPacketLossFraction),
+ CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
{kNumControllers - 2, kNumControllers - 1, 0, 1});
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs));
// Then let network metrics move a little towards the other controller.
- CheckControllersOrder(
- &states, rtc::Optional<int>(kBandwidthBps - kMinBandwithChangeBps - 1),
- rtc::Optional<float>(kPacketLossFraction),
- {kNumControllers - 2, kNumControllers - 1, 1, 0});
+ CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps - 1,
+ kPacketLossFraction,
+ {kNumControllers - 2, kNumControllers - 1, 1, 0});
}
TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
@@ -193,16 +188,14 @@
2.0f;
// Set network metrics to be in the middle between the characteristic points
// of two controllers.
- CheckControllersOrder(&states, rtc::Optional<int>(kBandwidthBps),
- rtc::Optional<float>(kPacketLossFraction),
+ CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
{kNumControllers - 2, kNumControllers - 1, 0, 1});
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs));
// Then let network metrics move a little towards the other controller.
- CheckControllersOrder(
- &states, rtc::Optional<int>(kBandwidthBps - kMinBandwithChangeBps + 1),
- rtc::Optional<float>(kPacketLossFraction),
- {kNumControllers - 2, kNumControllers - 1, 0, 1});
+ CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps + 1,
+ kPacketLossFraction,
+ {kNumControllers - 2, kNumControllers - 1, 0, 1});
}
#if WEBRTC_ENABLE_PROTOBUF
@@ -312,24 +305,19 @@
// initial values.
switch (expected_types[i]) {
case ControllerType::FEC:
- EXPECT_EQ(rtc::Optional<bool>(kInitialFecEnabled),
- encoder_config.enable_fec);
+ EXPECT_EQ(kInitialFecEnabled, encoder_config.enable_fec);
break;
case ControllerType::CHANNEL:
- EXPECT_EQ(rtc::Optional<size_t>(kIntialChannelsToEncode),
- encoder_config.num_channels);
+ EXPECT_EQ(kIntialChannelsToEncode, encoder_config.num_channels);
break;
case ControllerType::DTX:
- EXPECT_EQ(rtc::Optional<bool>(kInitialDtxEnabled),
- encoder_config.enable_dtx);
+ EXPECT_EQ(kInitialDtxEnabled, encoder_config.enable_dtx);
break;
case ControllerType::FRAME_LENGTH:
- EXPECT_EQ(rtc::Optional<int>(kInitialFrameLengthMs),
- encoder_config.frame_length_ms);
+ EXPECT_EQ(kInitialFrameLengthMs, encoder_config.frame_length_ms);
break;
case ControllerType::BIT_RATE:
- EXPECT_EQ(rtc::Optional<int>(kInitialBitrateBps),
- encoder_config.bitrate_bps);
+ EXPECT_EQ(kInitialBitrateBps, encoder_config.bitrate_bps);
}
}
}
@@ -448,10 +436,8 @@
auto states = CreateControllerManager(config_string);
Controller::NetworkMetrics metrics;
- metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(kChracteristicBandwithBps[0]);
- metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(kChracteristicPacketLossFraction[0]);
+ metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[0];
+ metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[0];
auto controllers = states.controller_manager->GetSortedControllers(metrics);
CheckControllersOrder(controllers,
@@ -460,10 +446,8 @@
ControllerType::CHANNEL, ControllerType::DTX,
ControllerType::BIT_RATE});
- metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(kChracteristicBandwithBps[1]);
- metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(kChracteristicPacketLossFraction[1]);
+ metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[1];
+ metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[1];
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs - 1));
controllers = states.controller_manager->GetSortedControllers(metrics);
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
index 7c7d6ad..cbfea95 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
@@ -44,7 +44,7 @@
dtx_enabled_ = true;
}
}
- config->enable_dtx = rtc::Optional<bool>(dtx_enabled_);
+ config->enable_dtx = dtx_enabled_;
}
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index 2c58249..e38e65d 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -39,7 +39,7 @@
}
AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
- EXPECT_EQ(rtc::Optional<bool>(expected_dtx_enabled), config.enable_dtx);
+ EXPECT_EQ(expected_dtx_enabled, config.enable_dtx);
}
} // namespace
@@ -47,43 +47,35 @@
TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr bool kInitialDtxEnabled = true;
auto controller = CreateController(kInitialDtxEnabled);
- CheckDecision(controller.get(), rtc::Optional<int>(), kInitialDtxEnabled);
+ CheckDecision(controller.get(), rtc::nullopt, kInitialDtxEnabled);
}
TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
auto controller = CreateController(false);
- CheckDecision(controller.get(), rtc::Optional<int>(kDtxEnablingBandwidthBps),
- true);
+ CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
}
TEST(DtxControllerTest, TurnOffDtxForHighUplinkBandwidth) {
auto controller = CreateController(true);
- CheckDecision(controller.get(), rtc::Optional<int>(kDtxDisablingBandwidthBps),
- false);
+ CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
}
TEST(DtxControllerTest, MaintainDtxOffForMediumUplinkBandwidth) {
auto controller = CreateController(false);
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps),
- false);
+ CheckDecision(controller.get(), kMediumBandwidthBps, false);
}
TEST(DtxControllerTest, MaintainDtxOnForMediumUplinkBandwidth) {
auto controller = CreateController(true);
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps),
- true);
+ CheckDecision(controller.get(), kMediumBandwidthBps, true);
}
TEST(DtxControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
auto controller = CreateController(false);
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps),
- false);
- CheckDecision(controller.get(), rtc::Optional<int>(kDtxEnablingBandwidthBps),
- true);
- CheckDecision(controller.get(), rtc::Optional<int>(kMediumBandwidthBps),
- true);
- CheckDecision(controller.get(), rtc::Optional<int>(kDtxDisablingBandwidthBps),
- false);
+ CheckDecision(controller.get(), kMediumBandwidthBps, false);
+ CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
+ CheckDecision(controller.get(), kMediumBandwidthBps, true);
+ CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
}
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
index ee8249a..df97594 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
@@ -51,13 +51,12 @@
state.event_log_writer.reset(new EventLogWriter(
state.event_log.get(), kMinBitrateChangeBps, kMinBitrateChangeFraction,
kMinPacketLossChangeFraction));
- state.runtime_config.bitrate_bps = rtc::Optional<int>(kHighBitrateBps);
- state.runtime_config.frame_length_ms = rtc::Optional<int>(kFrameLengthMs);
- state.runtime_config.uplink_packet_loss_fraction =
- rtc::Optional<float>(kPacketLossFraction);
- state.runtime_config.enable_fec = rtc::Optional<bool>(kEnableFec);
- state.runtime_config.enable_dtx = rtc::Optional<bool>(kEnableDtx);
- state.runtime_config.num_channels = rtc::Optional<size_t>(kNumChannels);
+ state.runtime_config.bitrate_bps = kHighBitrateBps;
+ state.runtime_config.frame_length_ms = kFrameLengthMs;
+ state.runtime_config.uplink_packet_loss_fraction = kPacketLossFraction;
+ state.runtime_config.enable_fec = kEnableFec;
+ state.runtime_config.enable_dtx = kEnableDtx;
+ state.runtime_config.num_channels = kNumChannels;
return state;
}
} // namespace
@@ -86,7 +85,7 @@
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.enable_fec = rtc::Optional<bool>(!kEnableFec);
+ state.runtime_config.enable_fec = !kEnableFec;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -100,7 +99,7 @@
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.enable_dtx = rtc::Optional<bool>(!kEnableDtx);
+ state.runtime_config.enable_dtx = !kEnableDtx;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -114,7 +113,7 @@
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.num_channels = rtc::Optional<size_t>(kNumChannels + 1);
+ state.runtime_config.num_channels = kNumChannels + 1;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -128,7 +127,7 @@
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.frame_length_ms = rtc::Optional<int>(20);
+ state.runtime_config.frame_length_ms = 20;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -141,8 +140,7 @@
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps - 1);
+ state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps - 1;
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
}
@@ -156,8 +154,7 @@
// min change rule. We make sure that the min change rule applies.
RTC_DCHECK_GT(kHighBitrateBps * kMinBitrateChangeFraction,
kMinBitrateChangeBps);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps);
+ state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -166,15 +163,15 @@
TEST(EventLogWriterTest, LogMinBitrateChangeFractionOnLowBitrateChange) {
auto state = CreateEventLogWriter();
- state.runtime_config.bitrate_bps = rtc::Optional<int>(kLowBitrateBps);
+ state.runtime_config.bitrate_bps = kLowBitrateBps;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
// At high bitrate, the min change rule requires a larger change than the min
// fraction rule. We make sure that the min fraction rule applies.
- state.runtime_config.bitrate_bps = rtc::Optional<int>(
- kLowBitrateBps + kLowBitrateBps * kMinBitrateChangeFraction);
+ state.runtime_config.bitrate_bps =
+ kLowBitrateBps + kLowBitrateBps * kMinBitrateChangeFraction;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -187,9 +184,9 @@
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
+ state.runtime_config.uplink_packet_loss_fraction =
kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction -
- 0.001f);
+ 0.001f;
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
}
@@ -199,8 +196,8 @@
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
- kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction);
+ state.runtime_config.uplink_packet_loss_fraction =
+ kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -213,9 +210,9 @@
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
- kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction);
- state.runtime_config.frame_length_ms = rtc::Optional<int>(20);
+ state.runtime_config.uplink_packet_loss_fraction =
+ kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
+ state.runtime_config.frame_length_ms = 20;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
@@ -228,14 +225,13 @@
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps);
+ state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
EXPECT_CALL(*state.event_log,
LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
.Times(1);
for (int bitrate_bps = kHighBitrateBps;
bitrate_bps <= kHighBitrateBps + kMinBitrateChangeBps; bitrate_bps++) {
- state.runtime_config.bitrate_bps = rtc::Optional<int>(bitrate_bps);
+ state.runtime_config.bitrate_bps = bitrate_bps;
state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
}
}
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 80cc695..62f356d 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -21,9 +21,7 @@
namespace {
class NullSmoothingFilter final : public SmoothingFilter {
public:
- void AddSample(float sample) override {
- last_sample_ = rtc::Optional<float>(sample);
- }
+ void AddSample(float sample) override { last_sample_ = sample; }
rtc::Optional<float> GetAverage() override { return last_sample_; }
@@ -85,10 +83,9 @@
fec_enabled_ = fec_enabled_ ? !FecDisablingDecision(packet_loss)
: FecEnablingDecision(packet_loss);
- config->enable_fec = rtc::Optional<bool>(fec_enabled_);
+ config->enable_fec = fec_enabled_;
- config->uplink_packet_loss_fraction =
- rtc::Optional<float>(packet_loss ? *packet_loss : 0.0);
+ config->uplink_packet_loss_fraction = packet_loss ? *packet_loss : 0.0;
}
bool FecControllerPlrBased::FecEnablingDecision(
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index 41d959d..8636aa9 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -95,17 +95,10 @@
states->controller->UpdateNetworkMetrics(network_metrics);
// This is called during CheckDecision().
EXPECT_CALL(*states->packet_loss_smoother, GetAverage())
- .WillOnce(Return(rtc::Optional<float>(*uplink_packet_loss)));
+ .WillOnce(Return(*uplink_packet_loss));
}
}
-void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
- int uplink_bandwidth_bps,
- float uplink_packet_loss) {
- UpdateNetworkMetrics(states, rtc::Optional<int>(uplink_bandwidth_bps),
- rtc::Optional<float>(uplink_packet_loss));
-}
-
// Checks that the FEC decision and |uplink_packet_loss_fraction| given by
// |states->controller->MakeDecision| matches |expected_enable_fec| and
// |expected_uplink_packet_loss_fraction|, respectively.
@@ -114,8 +107,8 @@
float expected_uplink_packet_loss_fraction) {
AudioEncoderRuntimeConfig config;
states->controller->MakeDecision(&config);
- EXPECT_EQ(rtc::Optional<bool>(expected_enable_fec), config.enable_fec);
- EXPECT_EQ(rtc::Optional<float>(expected_uplink_packet_loss_fraction),
+ EXPECT_EQ(expected_enable_fec, config.enable_fec);
+ EXPECT_EQ(expected_uplink_packet_loss_fraction,
config.uplink_packet_loss_fraction);
}
@@ -138,8 +131,7 @@
kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
kEnablingPacketLossAtLowBw + kEpsilon}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, rtc::Optional<int>(),
- rtc::Optional<float>(packet_loss));
+ UpdateNetworkMetrics(&states, rtc::nullopt, packet_loss);
CheckDecision(&states, initial_fec_enabled, packet_loss);
}
}
@@ -154,8 +146,7 @@
kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, rtc::Optional<int>(bandwidth),
- rtc::Optional<float>());
+ UpdateNetworkMetrics(&states, bandwidth, rtc::nullopt);
CheckDecision(&states, initial_fec_enabled, 0.0);
}
}
@@ -178,12 +169,10 @@
// audio_network_adaptor_impl.cc.
auto states = CreateFecControllerPlrBased(false);
Controller::NetworkMetrics network_metrics;
- network_metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(kEnablingBandwidthHigh);
- network_metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(kEnablingPacketLossAtHighBw);
+ network_metrics.uplink_bandwidth_bps = kEnablingBandwidthHigh;
+ network_metrics.uplink_packet_loss_fraction = kEnablingPacketLossAtHighBw;
EXPECT_CALL(*states.packet_loss_smoother, GetAverage())
- .WillOnce(Return(rtc::Optional<float>(kEnablingPacketLossAtHighBw)));
+ .WillOnce(Return(kEnablingPacketLossAtHighBw));
states.controller->UpdateNetworkMetrics(network_metrics);
CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
}
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
index eb56ea0..c8cfd31 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
@@ -48,9 +48,9 @@
fec_enabled_ = fec_enabled_ ? !FecDisablingDecision() : FecEnablingDecision();
- config->enable_fec = rtc::Optional<bool>(fec_enabled_);
- config->uplink_packet_loss_fraction = rtc::Optional<float>(
- uplink_recoverable_packet_loss_ ? *uplink_recoverable_packet_loss_ : 0.0);
+ config->enable_fec = fec_enabled_;
+ config->uplink_packet_loss_fraction =
+ uplink_recoverable_packet_loss_ ? *uplink_recoverable_packet_loss_ : 0.0;
}
bool FecControllerRplrBased::FecEnablingDecision() const {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
index f20122f..0fc003b 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
@@ -49,11 +49,9 @@
std::mt19937 generator(rd());
std::uniform_real_distribution<> distribution(0, 1);
- if (distribution(generator) < 0.2) {
- return rtc::Optional<float>();
- } else {
- return rtc::Optional<float>(distribution(generator));
- }
+ return (distribution(generator) < 0.2)
+ ? rtc::nullopt
+ : rtc::Optional<float>(distribution(generator));
}
std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
@@ -108,13 +106,6 @@
uplink_recoveralbe_packet_loss);
}
-void UpdateNetworkMetrics(FecControllerRplrBased* controller,
- int uplink_bandwidth_bps,
- float uplink_recoveralbe_packet_loss) {
- UpdateNetworkMetrics(controller, rtc::Optional<int>(uplink_bandwidth_bps),
- rtc::Optional<float>(uplink_recoveralbe_packet_loss));
-}
-
// Checks that the FEC decision and |uplink_packet_loss_fraction| given by
// |states->controller->MakeDecision| matches |expected_enable_fec| and
// |expected_uplink_packet_loss_fraction|, respectively.
@@ -157,8 +148,8 @@
kEnablingRecoverablePacketLossAtHighBw,
kEnablingRecoverablePacketLossAtHighBw + kEpsilon}) {
auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(),
- rtc::Optional<float>(recoverable_packet_loss));
+ UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ recoverable_packet_loss);
CheckDecision(controller.get(), initial_fec_enabled,
recoverable_packet_loss);
}
@@ -174,8 +165,7 @@
kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(bandwidth),
- rtc::Optional<float>());
+ UpdateNetworkMetrics(controller.get(), bandwidth, rtc::nullopt);
CheckDecision(controller.get(), initial_fec_enabled, 0.0);
}
}
@@ -198,12 +188,10 @@
// audio_network_adaptor_impl.cc.
auto controller = CreateFecControllerRplrBased(false);
Controller::NetworkMetrics network_metrics;
- network_metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(kEnablingBandwidthHigh);
- network_metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(GetRandomProbabilityOrUnknown());
+ network_metrics.uplink_bandwidth_bps = kEnablingBandwidthHigh;
+ network_metrics.uplink_packet_loss_fraction = GetRandomProbabilityOrUnknown();
network_metrics.uplink_recoverable_packet_loss_fraction =
- rtc::Optional<float>(kEnablingRecoverablePacketLossAtHighBw);
+ kEnablingRecoverablePacketLossAtHighBw;
controller->UpdateNetworkMetrics(network_metrics);
CheckDecision(controller.get(), true, kEnablingRecoverablePacketLossAtHighBw);
}
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index 5ae78b0..6c3cae0 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -82,7 +82,7 @@
prev_decision_increase_ = false;
}
config->last_fl_change_increase = prev_decision_increase_;
- config->frame_length_ms = rtc::Optional<int>(*frame_length_ms_);
+ config->frame_length_ms = *frame_length_ms_;
}
FrameLengthController::Config::FrameLengthChange::FrameLengthChange(
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 4c96c96..1f98447 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -105,8 +105,7 @@
int expected_frame_length_ms) {
AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
- EXPECT_EQ(rtc::Optional<int>(expected_frame_length_ms),
- config.frame_length_ms);
+ EXPECT_EQ(expected_frame_length_ms, config.frame_length_ms);
}
} // namespace
@@ -114,18 +113,17 @@
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkBandwidth) {
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
- UpdateNetworkMetrics(
- controller.get(), rtc::Optional<int>(kFl60msTo20msBandwidthBps),
- rtc::Optional<float>(), rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+ rtc::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -145,10 +143,8 @@
// 3. FEC is not decided ON.
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
- UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kMediumBandwidthBps),
- rtc::Optional<float>(kMediumPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+ kMediumPacketLossFraction, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -160,10 +156,9 @@
// 3. FEC is not decided or OFF.
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
- UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl20msTo60msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -173,9 +168,9 @@
// We set packet loss fraction to kFlDecreasingPacketLossFraction, which
// should have prevented frame length to increase, if the uplink bandwidth
// was not this low.
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(VeryLowBitrate(20)),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -185,9 +180,9 @@
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have caused the frame length to decrease, if the uplink bandwidth was not
// this low.
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(VeryLowBitrate(20)),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -202,10 +197,8 @@
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
Controller::NetworkMetrics network_metrics;
- network_metrics.uplink_bandwidth_bps =
- rtc::Optional<int>(kFl20msTo60msBandwidthBps);
- network_metrics.uplink_packet_loss_fraction =
- rtc::Optional<float>(kFlIncreasingPacketLossFraction);
+ network_metrics.uplink_bandwidth_bps = kFl20msTo60msBandwidthBps;
+ network_metrics.uplink_packet_loss_fraction = kFlIncreasingPacketLossFraction;
controller->UpdateNetworkMetrics(network_metrics);
CheckDecision(controller.get(), 60);
}
@@ -217,9 +210,9 @@
// Use a low uplink bandwidth and a low uplink packet loss fraction that would
// cause frame length to increase if receiver frame length included 60ms.
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl20msTo60msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl20msTo60msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -227,9 +220,9 @@
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kMediumBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kMediumBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -239,9 +232,9 @@
// Use a low uplink bandwidth that would cause frame length to increase if
// uplink packet loss fraction was low.
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl20msTo60msBandwidthBps),
- rtc::Optional<float>(kMediumPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl20msTo60msBandwidthBps,
+ kMediumPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -249,9 +242,9 @@
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60, 120}, 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -259,14 +252,16 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
- UpdateNetworkMetrics(
- controller.get(), rtc::Optional<int>(kFl60msTo20msBandwidthBps),
- rtc::Optional<float>(), rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(),
+ kFl60msTo20msBandwidthBps,
+ rtc::nullopt,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(
- controller.get(), rtc::Optional<int>(kFl60msTo20msBandwidthBps),
- rtc::Optional<float>(), rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(),
+ kFl60msTo20msBandwidthBps,
+ rtc::nullopt,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -274,14 +269,14 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -291,9 +286,9 @@
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have caused the frame length to decrease, if the uplink bandwidth was not
// this low.
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(VeryLowBitrate(60)),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
}
@@ -303,9 +298,9 @@
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have prevented frame length to increase, if the uplink bandwidth was not
// this low.
- UpdateNetworkMetrics(controller.get(), rtc::Optional<int>(VeryLowBitrate(60)),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
}
@@ -317,14 +312,14 @@
{20, 60, 120}, 20);
// It takes two steps for frame length to go from 20ms to 120ms.
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
}
@@ -332,14 +327,14 @@
auto controller =
CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(), {20, 60}, 20);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
}
@@ -347,39 +342,39 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 20);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kMediumBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kMediumBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl20msTo60msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl20msTo60msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kMediumPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kMediumPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl60msTo120msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl60msTo120msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 120);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kFl120msTo60msBandwidthBps),
- rtc::Optional<float>(kFlIncreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kFl120msTo60msBandwidthBps,
+ kFlIncreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(),
- rtc::Optional<int>(kMediumBandwidthBps),
- rtc::Optional<float>(kFlDecreasingPacketLossFraction),
- rtc::Optional<size_t>(kOverheadBytesPerPacket));
+ kMediumBandwidthBps,
+ kFlDecreasingPacketLossFraction,
+ kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 5779625..303529e 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -222,7 +222,7 @@
CreateCng(MakeCngConfig());
EXPECT_CALL(*mock_encoder_,
OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
- cng_->OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>());
+ cng_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
}
TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index 8c35b64..341336e 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -35,9 +35,9 @@
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
if (ret < 0)
- return rtc::Optional<DecodeResult>();
+ return rtc::nullopt;
- return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
+ return DecodeResult{static_cast<size_t>(ret), speech_type};
}
std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index a48c7db..3d10b6f 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -51,9 +51,9 @@
}
if (ret < 0)
- return rtc::Optional<DecodeResult>();
+ return rtc::nullopt;
- return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
+ return DecodeResult{static_cast<size_t>(ret), speech_type};
}
private:
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index d5358bb..5a5ac34 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -108,9 +108,10 @@
rtc::Optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
const std::string& param) {
auto it = format.parameters.find(param);
- return (it == format.parameters.end())
- ? rtc::Optional<std::string>()
- : rtc::Optional<std::string>(it->second);
+ if (it == format.parameters.end())
+ return rtc::nullopt;
+
+ return it->second;
}
template <typename T>
@@ -255,9 +256,9 @@
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
- return rtc::Optional<AudioCodecInfo>(info);
+ return info;
}
- return rtc::Optional<AudioCodecInfo>();
+ return rtc::nullopt;
}
AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
@@ -265,7 +266,7 @@
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
- config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
+ config.bitrate_bps = codec_inst.rate;
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
@@ -277,7 +278,7 @@
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
- return rtc::Optional<AudioEncoderOpusConfig>();
+ return rtc::nullopt;
}
AudioEncoderOpusConfig config;
@@ -287,9 +288,9 @@
config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
- config.bitrate_bps = rtc::Optional<int>(
+ config.bitrate_bps =
CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
- GetFormatParameter(format, "maxaveragebitrate")));
+ GetFormatParameter(format, "maxaveragebitrate"));
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
@@ -309,7 +310,7 @@
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config.supported_frame_lengths_ms);
RTC_DCHECK(config.IsOk());
- return rtc::Optional<AudioEncoderOpusConfig>(config);
+ return config;
}
rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
@@ -321,11 +322,11 @@
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
- return rtc::Optional<int>();
+ return rtc::nullopt;
} else {
- return rtc::Optional<int>(bitrate_bps <= config.complexity_threshold_bps
- ? config.low_rate_complexity
- : config.complexity);
+ return bitrate_bps <= config.complexity_threshold_bps
+ ? config.low_rate_complexity
+ : config.complexity;
}
}
@@ -552,8 +553,7 @@
audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
ApplyAudioNetworkAdaptor();
} else {
- overhead_bytes_per_packet_ =
- rtc::Optional<size_t>(overhead_bytes_per_packet);
+ overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
}
@@ -707,9 +707,9 @@
}
void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
- config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
+ config_.bitrate_bps = rtc::SafeClamp<int>(
bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
- AudioEncoderOpusConfig::kMaxBitrateBps));
+ AudioEncoderOpusConfig::kMaxBitrateBps);
RTC_DCHECK(config_.IsOk());
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
const auto new_complexity = GetNewComplexity(config_);
@@ -759,7 +759,7 @@
rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
- bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
+ bitrate_smoother_last_update_time_ = now_ms;
}
}
}
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 5a83a4b..868de8c 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -38,7 +38,7 @@
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
- config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
+ config.bitrate_bps = codec_inst.rate;
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
@@ -102,13 +102,12 @@
constexpr size_t kNumChannels = 1;
constexpr float kPacketLossFraction = 0.1f;
AudioEncoderRuntimeConfig config;
- config.bitrate_bps = rtc::Optional<int>(kBitrate);
- config.frame_length_ms = rtc::Optional<int>(kFrameLength);
- config.enable_fec = rtc::Optional<bool>(kEnableFec);
- config.enable_dtx = rtc::Optional<bool>(kEnableDtx);
- config.num_channels = rtc::Optional<size_t>(kNumChannels);
- config.uplink_packet_loss_fraction =
- rtc::Optional<float>(kPacketLossFraction);
+ config.bitrate_bps = kBitrate;
+ config.frame_length_ms = kFrameLength;
+ config.enable_fec = kEnableFec;
+ config.enable_dtx = kEnableDtx;
+ config.num_channels = kNumChannels;
+ config.uplink_packet_loss_fraction = kPacketLossFraction;
return config;
}
@@ -201,24 +200,20 @@
const int kMinBitrateBps = 6000;
const int kMaxBitrateBps = 510000;
// Set a too low bitrate.
- states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, rtc::nullopt);
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set a too high bitrate.
- states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1, rtc::nullopt);
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
// Set the minimum rate.
- states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps, rtc::nullopt);
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set the maximum rate.
- states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, rtc::nullopt);
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
// Set rates from kMaxBitrateBps up to 32000 bps.
for (int rate = kMinBitrateBps; rate <= 32000; rate += 1000) {
- states.encoder->OnReceivedUplinkBandwidth(rate, rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
}
}
@@ -329,8 +324,8 @@
EXPECT_CALL(*states.mock_bitrate_smoother,
SetTimeConstantMs(kProbingIntervalMs * 4));
EXPECT_CALL(*states.mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
- states.encoder->OnReceivedUplinkBandwidth(
- kTargetAudioBitrate, rtc::Optional<int64_t>(kProbingIntervalMs));
+ states.encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
+ kProbingIntervalMs);
CheckEncoderRuntimeConfig(states.encoder.get(), config);
}
@@ -401,7 +396,7 @@
auto states = CreateCodec(2);
states.encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
- rtc::Optional<int64_t>());
+ rtc::nullopt);
// Since |OnReceivedOverhead| has not been called, the codec bitrate should
// not change.
@@ -418,8 +413,7 @@
states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
constexpr int kTargetBitrateBps = 40000;
- states.encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, rtc::nullopt);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
EXPECT_EQ(kTargetBitrateBps -
@@ -445,16 +439,14 @@
// subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.
int target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMinBitrateBps - 1;
- states.encoder->OnReceivedUplinkBandwidth(target_bitrate,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set a target rate that is greater than |kMaxBitrateBps| when overhead is
// subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
- states.encoder->OnReceivedUplinkBandwidth(target_bitrate,
- rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
}
@@ -465,24 +457,20 @@
config.complexity = 6;
// Bitrate within hysteresis window. Expect empty output.
- config.bitrate_bps = rtc::Optional<int>(12500);
- EXPECT_EQ(rtc::Optional<int>(),
- AudioEncoderOpusImpl::GetNewComplexity(config));
+ config.bitrate_bps = 12500;
+ EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate below hysteresis window. Expect higher complexity.
- config.bitrate_bps = rtc::Optional<int>(10999);
- EXPECT_EQ(rtc::Optional<int>(8),
- AudioEncoderOpusImpl::GetNewComplexity(config));
+ config.bitrate_bps = 10999;
+ EXPECT_EQ(8, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate within hysteresis window. Expect empty output.
- config.bitrate_bps = rtc::Optional<int>(12500);
- EXPECT_EQ(rtc::Optional<int>(),
- AudioEncoderOpusImpl::GetNewComplexity(config));
+ config.bitrate_bps = 12500;
+ EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate above hysteresis window. Expect lower complexity.
- config.bitrate_bps = rtc::Optional<int>(14001);
- EXPECT_EQ(rtc::Optional<int>(6),
- AudioEncoderOpusImpl::GetNewComplexity(config));
+ config.bitrate_bps = 14001;
+ EXPECT_EQ(6, AudioEncoderOpusImpl::GetNewComplexity(config));
}
TEST(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
@@ -512,7 +500,7 @@
audio.fill(0);
rtc::Buffer encoded;
EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
- .WillOnce(Return(rtc::Optional<float>(50000)));
+ .WillOnce(Return(50000));
EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(50000));
states.encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
@@ -527,7 +515,7 @@
// Update when it is time to update.
EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
- .WillOnce(Return(rtc::Optional<float>(40000)));
+ .WillOnce(Return(40000));
EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(40000));
states.fake_clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
states.encoder->Encode(
@@ -544,7 +532,7 @@
rtc::Buffer encoded;
uint32_t rtp_timestamp = 12345; // Just a number not important to this test.
- states.encoder->OnReceivedUplinkBandwidth(0, rtc::Optional<int64_t>());
+ states.encoder->OnReceivedUplinkBandwidth(0, rtc::nullopt);
for (int packet_index = 0; packet_index < kNumPacketsToEncode;
packet_index++) {
// Make sure we are not encoding before we have enough data for
diff --git a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
index 6d9e911..4c9174d 100644
--- a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -64,11 +64,11 @@
AudioEncoderOpusConfig config;
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased.
- config.bitrate_bps = rtc::Optional<int>(11000 - 1);
+ config.bitrate_bps = 11000 - 1;
config.low_rate_complexity = 9;
int64_t runtime_10999bps = RunComplexityTest(config);
- config.bitrate_bps = rtc::Optional<int>(15500);
+ config.bitrate_bps = 15500;
int64_t runtime_15500bps = RunComplexityTest(config);
test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on",
@@ -85,10 +85,10 @@
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased (but not in this test since complexity adaptation is
// disabled).
- config.bitrate_bps = rtc::Optional<int>(11000 - 1);
+ config.bitrate_bps = 11000 - 1;
int64_t runtime_10999bps = RunComplexityTest(config);
- config.bitrate_bps = rtc::Optional<int>(15500);
+ config.bitrate_bps = 15500;
int64_t runtime_15500bps = RunComplexityTest(config);
test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 8409b82..8c63796 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -101,7 +101,7 @@
TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
EXPECT_CALL(*mock_encoder_,
OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
- red_->OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>());
+ red_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
}
TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index ae164fd..3181d6f 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -462,7 +462,7 @@
namespace {
int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
- audio_encoder->OnReceivedUplinkBandwidth(rate, rtc::Optional<int64_t>());
+ audio_encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
return audio_encoder->GetTargetBitrate();
}
void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 743ca87..d289b45 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -104,10 +104,9 @@
const int sample_rate_hz = format.clockrate_hz;
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000);
- return rtc::Optional<DecoderDatabase::DecoderInfo::CngDecoder>(
- {sample_rate_hz});
+ return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
} else {
- return rtc::Optional<CngDecoder>();
+ return rtc::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/neteq_decoder_enum.cc b/modules/audio_coding/neteq/neteq_decoder_enum.cc
index 2c61b0a..8d66c2a 100644
--- a/modules/audio_coding/neteq/neteq_decoder_enum.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_enum.cc
@@ -18,71 +18,67 @@
rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
switch (nd) {
case NetEqDecoder::kDecoderPCMu:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 1));
+ return SdpAudioFormat("pcmu", 8000, 1);
case NetEqDecoder::kDecoderPCMa:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 1));
+ return SdpAudioFormat("pcma", 8000, 1);
case NetEqDecoder::kDecoderPCMu_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 2));
+ return SdpAudioFormat("pcmu", 8000, 2);
case NetEqDecoder::kDecoderPCMa_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 2));
+ return SdpAudioFormat("pcma", 8000, 2);
case NetEqDecoder::kDecoderILBC:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("ilbc", 8000, 1));
+ return SdpAudioFormat("ilbc", 8000, 1);
case NetEqDecoder::kDecoderISAC:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 16000, 1));
+ return SdpAudioFormat("isac", 16000, 1);
case NetEqDecoder::kDecoderISACswb:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 32000, 1));
+ return SdpAudioFormat("isac", 32000, 1);
case NetEqDecoder::kDecoderPCM16B:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 1));
+ return SdpAudioFormat("l16", 8000, 1);
case NetEqDecoder::kDecoderPCM16Bwb:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 1));
+ return SdpAudioFormat("l16", 16000, 1);
case NetEqDecoder::kDecoderPCM16Bswb32kHz:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 1));
+ return SdpAudioFormat("l16", 32000, 1);
case NetEqDecoder::kDecoderPCM16Bswb48kHz:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 1));
+ return SdpAudioFormat("l16", 48000, 1);
case NetEqDecoder::kDecoderPCM16B_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 2));
+ return SdpAudioFormat("l16", 8000, 2);
case NetEqDecoder::kDecoderPCM16Bwb_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 2));
+ return SdpAudioFormat("l16", 16000, 2);
case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 2));
+ return SdpAudioFormat("l16", 32000, 2);
case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 2));
+ return SdpAudioFormat("l16", 48000, 2);
case NetEqDecoder::kDecoderPCM16B_5ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 5));
+ return SdpAudioFormat("l16", 8000, 5);
case NetEqDecoder::kDecoderG722:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 1));
+ return SdpAudioFormat("g722", 8000, 1);
case NetEqDecoder::kDecoderG722_2ch:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 2));
+ return SdpAudioFormat("g722", 8000, 2);
case NetEqDecoder::kDecoderOpus:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("opus", 48000, 2));
+ return SdpAudioFormat("opus", 48000, 2);
case NetEqDecoder::kDecoderOpus_2ch:
- return rtc::Optional<SdpAudioFormat>(
- SdpAudioFormat("opus", 48000, 2,
- std::map<std::string, std::string>{{"stereo", "1"}}));
+ return SdpAudioFormat(
+ "opus", 48000, 2,
+ std::map<std::string, std::string>{{"stereo", "1"}});
case NetEqDecoder::kDecoderRED:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("red", 8000, 1));
+ return SdpAudioFormat("red", 8000, 1);
case NetEqDecoder::kDecoderAVT:
- return rtc::Optional<SdpAudioFormat>(
- SdpAudioFormat("telephone-event", 8000, 1));
+ return SdpAudioFormat("telephone-event", 8000, 1);
case NetEqDecoder::kDecoderAVT16kHz:
- return rtc::Optional<SdpAudioFormat>(
- SdpAudioFormat("telephone-event", 16000, 1));
+ return SdpAudioFormat("telephone-event", 16000, 1);
case NetEqDecoder::kDecoderAVT32kHz:
- return rtc::Optional<SdpAudioFormat>(
- SdpAudioFormat("telephone-event", 32000, 1));
+ return SdpAudioFormat("telephone-event", 32000, 1);
case NetEqDecoder::kDecoderAVT48kHz:
- return rtc::Optional<SdpAudioFormat>(
- SdpAudioFormat("telephone-event", 48000, 1));
+ return SdpAudioFormat("telephone-event", 48000, 1);
case NetEqDecoder::kDecoderCNGnb:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 8000, 1));
+ return SdpAudioFormat("cn", 8000, 1);
case NetEqDecoder::kDecoderCNGwb:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 16000, 1));
+ return SdpAudioFormat("cn", 16000, 1);
case NetEqDecoder::kDecoderCNGswb32kHz:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 32000, 1));
+ return SdpAudioFormat("cn", 32000, 1);
case NetEqDecoder::kDecoderCNGswb48kHz:
- return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 48000, 1));
+ return SdpAudioFormat("cn", 48000, 1);
default:
- return rtc::Optional<SdpAudioFormat>();
+ return rtc::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 089f6ca..14f2fc7 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -423,10 +423,9 @@
// We don't have a valid RTP timestamp until we have decoded our first
// RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
// which is indicated by returning an empty value.
- return rtc::Optional<uint32_t>();
+ return rtc::nullopt;
}
- return rtc::Optional<uint32_t>(
- timestamp_scaler_->ToExternal(playout_timestamp_));
+ return timestamp_scaler_->ToExternal(playout_timestamp_);
}
int NetEqImpl::last_output_sample_rate_hz() const {
@@ -439,7 +438,7 @@
const DecoderDatabase::DecoderInfo* di =
decoder_database_->GetDecoderInfo(payload_type);
if (!di) {
- return rtc::Optional<CodecInst>();
+ return rtc::nullopt;
}
// Create a CodecInst with some fields set. The remaining fields are zeroed,
@@ -452,7 +451,7 @@
ci.plfreq = di->IsRed() ? 8000 : di->SampleRateHz();
AudioDecoder* const decoder = di->GetDecoder();
ci.channels = decoder ? decoder->Channels() : 1;
- return rtc::Optional<CodecInst>(ci);
+ return ci;
}
rtc::Optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
@@ -461,9 +460,9 @@
const DecoderDatabase::DecoderInfo* const di =
decoder_database_->GetDecoderInfo(payload_type);
if (!di) {
- return rtc::Optional<SdpAudioFormat>(); // Payload type not registered.
+ return rtc::nullopt; // Payload type not registered.
}
- return rtc::Optional<SdpAudioFormat>(di->GetFormat());
+ return di->GetFormat();
}
int NetEqImpl::SetTargetNumberOfChannels() {
@@ -2004,7 +2003,7 @@
stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
packet_list->push_back(std::move(*packet)); // Store packet in list.
- packet = rtc::Optional<Packet>(); // Ensure it's never used after the move.
+ packet = rtc::nullopt; // Ensure it's never used after the move.
// Check what packet is available next.
next_packet = packet_buffer_->PeekNextPacket();
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index cade064..2a21e51 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -348,9 +348,8 @@
.Times(1);
EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _))
.Times(2)
- .WillRepeatedly(
- DoAll(SetArgPointee<2>(rtc::Optional<uint8_t>(kPayloadType)),
- WithArg<0>(Invoke(DeletePacketsAndReturnOk))));
+ .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType),
+ WithArg<0>(Invoke(DeletePacketsAndReturnOk))));
// SetArgPointee<2>(kPayloadType) means that the third argument (zero-based
// index) is a pointer, and the variable pointed to is set to kPayloadType.
// Also invoke the function DeletePacketsAndReturnOk to properly delete all
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 1173da2..334715f 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -53,12 +53,11 @@
if (decoded.size() >= output_size) {
memset(decoded.data(), 0,
sizeof(int16_t) * kPacketDuration * num_channels_);
- return rtc::Optional<DecodeResult>(
- {kPacketDuration * num_channels_, kSpeech});
+ return DecodeResult{kPacketDuration * num_channels_, kSpeech};
} else {
ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
<< decoded.size() << " vs. " << output_size << ")";
- return rtc::Optional<DecodeResult>();
+ return rtc::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 3005d43..dfffebd 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -139,12 +139,11 @@
if (*current_cng_rtp_payload_type &&
**current_cng_rtp_payload_type != packet.payload_type) {
// New CNG payload type implies new codec type.
- *current_rtp_payload_type = rtc::Optional<uint8_t>();
+ *current_rtp_payload_type = rtc::nullopt;
Flush();
flushed = true;
}
- *current_cng_rtp_payload_type =
- rtc::Optional<uint8_t>(packet.payload_type);
+ *current_cng_rtp_payload_type = packet.payload_type;
} else if (!decoder_database.IsDtmf(packet.payload_type)) {
// This must be speech.
if ((*current_rtp_payload_type &&
@@ -153,11 +152,11 @@
!EqualSampleRates(packet.payload_type,
**current_cng_rtp_payload_type,
decoder_database))) {
- *current_cng_rtp_payload_type = rtc::Optional<uint8_t>();
+ *current_cng_rtp_payload_type = rtc::nullopt;
Flush();
flushed = true;
}
- *current_rtp_payload_type = rtc::Optional<uint8_t>(packet.payload_type);
+ *current_rtp_payload_type = packet.payload_type;
}
int return_val = InsertPacket(std::move(packet), stats);
if (return_val == kFlushed) {
@@ -210,7 +209,7 @@
rtc::Optional<Packet> PacketBuffer::GetNextPacket() {
if (Empty()) {
// Buffer is empty.
- return rtc::Optional<Packet>();
+ return rtc::nullopt;
}
rtc::Optional<Packet> packet(std::move(buffer_.front()));
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 4beead6..0ddeb8a 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -188,9 +188,8 @@
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
- EXPECT_EQ(rtc::Optional<uint8_t>(0),
- current_pt); // Current payload type changed to 0.
- EXPECT_FALSE(current_cng_pt); // CNG payload type not changed.
+ EXPECT_EQ(0, current_pt); // Current payload type changed to 0.
+ EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -236,9 +235,8 @@
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); // Only the last packet.
- EXPECT_EQ(rtc::Optional<uint8_t>(1),
- current_pt); // Current payload type changed to 1.
- EXPECT_FALSE(current_cng_pt); // CNG payload type not changed.
+ EXPECT_EQ(1, current_pt); // Current payload type changed to 1.
+ EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -470,9 +468,8 @@
EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
ASSERT_TRUE(buffer.PeekNextPacket());
EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
- EXPECT_FALSE(current_pt); // Current payload type not set.
- EXPECT_EQ(rtc::Optional<uint8_t>(kCngPt),
- current_cng_pt); // CNG payload type set.
+ EXPECT_EQ(current_pt, rtc::nullopt); // Current payload type not set.
+ EXPECT_EQ(kCngPt, current_cng_pt); // CNG payload type set.
// Insert second packet, which is wide-band speech.
{
@@ -490,9 +487,8 @@
ASSERT_TRUE(buffer.PeekNextPacket());
EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
- EXPECT_EQ(rtc::Optional<uint8_t>(kSpeechPt),
- current_pt); // Current payload type set.
- EXPECT_FALSE(current_cng_pt); // CNG payload type reset.
+ EXPECT_EQ(kSpeechPt, current_pt); // Current payload type set.
+ EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type reset.
buffer.Flush(); // Clean up.
EXPECT_CALL(decoder_database, Die()); // Called when object is deleted.
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index d04e818..212b811 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -29,11 +29,11 @@
rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
RTC_DCHECK(packet_data_);
- return rtc::Optional<int64_t>(static_cast<int64_t>(packet_data_->time_ms));
+ return static_cast<int64_t>(packet_data_->time_ms);
}
rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
- return rtc::Optional<int64_t>(next_output_event_ms_);
+ return next_output_event_ms_;
}
std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
@@ -52,7 +52,7 @@
rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
RTC_DCHECK(packet_data_);
- return rtc::Optional<RTPHeader>(packet_data_->header);
+ return packet_data_->header;
}
void EncodeNetEqInput::CreatePacket() {
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index 6779e5e..393a46f 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -59,8 +59,7 @@
RTC_CHECK(input_->Seek(jump));
}
- next_timestamp_from_input_ =
- rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
+ next_timestamp_from_input_ = timestamp_to_decode + samples_to_decode;
uint32_t original_payload_size_bytes =
ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
index 72a539f..34e7a84 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -86,12 +86,11 @@
auto& it_timing = it->second;
RTC_CHECK(!it_timing.decode_get_audio_count)
<< "Decode time already written";
- it_timing.decode_get_audio_count = rtc::Optional<int64_t>(get_audio_count_);
+ it_timing.decode_get_audio_count = get_audio_count_;
RTC_CHECK(!it_timing.sync_delay_ms) << "Decode time already written";
- it_timing.sync_delay_ms = rtc::Optional<int64_t>(last_sync_buffer_ms_);
- it_timing.target_delay_ms = rtc::Optional<int>(neteq->TargetDelayMs());
- it_timing.current_delay_ms =
- rtc::Optional<int>(neteq->FilteredCurrentDelayMs());
+ it_timing.sync_delay_ms = last_sync_buffer_ms_;
+ it_timing.target_delay_ms = neteq->TargetDelayMs();
+ it_timing.current_delay_ms = neteq->FilteredCurrentDelayMs();
}
last_sample_rate_hz_ = audio_frame.sample_rate_hz_;
++get_audio_count_;
@@ -159,16 +158,16 @@
const float playout_ms = *timing.decode_get_audio_count * 10 +
get_audio_time_ms_[0] + *timing.sync_delay_ms -
offset_send_time_ms;
- playout_delay_ms->push_back(rtc::Optional<float>(playout_ms));
+ playout_delay_ms->push_back(playout_ms);
RTC_DCHECK(timing.target_delay_ms);
RTC_DCHECK(timing.current_delay_ms);
const float target =
playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
- target_delay_ms->push_back(rtc::Optional<float>(target));
+ target_delay_ms->push_back(target);
} else {
// This packet was never decoded. Mark target and playout delays as empty.
- playout_delay_ms->push_back(rtc::Optional<float>());
- target_delay_ms->push_back(rtc::Optional<float>());
+ playout_delay_ms->push_back(rtc::nullopt);
+ target_delay_ms->push_back(rtc::nullopt);
}
}
RTC_DCHECK(data_it == data_.end());
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 993ba04..88d9eb9 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -52,9 +52,9 @@
const auto b = NextOutputEventTime();
// Return the minimum of non-empty |a| and |b|, or empty if both are empty.
if (a) {
- return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a;
+ return b ? std::min(*a, *b) : a;
}
- return b ? b : rtc::Optional<int64_t>();
+ return b ? b : rtc::nullopt;
}
// Returns the next packet to be inserted into NetEq. The packet following the
diff --git a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
index 12a12c4..0741d7c 100644
--- a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -25,12 +25,12 @@
rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
return packet_
? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
- : rtc::Optional<int64_t>();
+ : rtc::nullopt;
}
rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
- : rtc::Optional<RTPHeader>();
+ : rtc::nullopt;
}
void NetEqPacketSourceInput::LoadNextPacket() {
@@ -78,7 +78,7 @@
*next_output_event_ms_ += kOutputPeriodMs;
}
if (!NextPacketTime()) {
- next_output_event_ms_ = rtc::Optional<int64_t>();
+ next_output_event_ms_ = rtc::nullopt;
}
}
@@ -97,14 +97,13 @@
}
rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
- return rtc::Optional<int64_t>(next_output_event_ms_);
+ return next_output_event_ms_;
}
void NetEqEventLogInput::AdvanceOutputEvent() {
- next_output_event_ms_ =
- rtc::Optional<int64_t>(source_->NextAudioOutputEventMs());
+ next_output_event_ms_ = source_->NextAudioOutputEventMs();
if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
- next_output_event_ms_ = rtc::Optional<int64_t>();
+ next_output_event_ms_ = rtc::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index a3e3413..6c846c0 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -34,7 +34,7 @@
rtc::Optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
return packet_
? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
- : rtc::Optional<int64_t>();
+ : rtc::nullopt;
}
rtc::Optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 7ea73fb..4c0f3bf 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -209,20 +209,20 @@
if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
- return rtc::Optional<int>(8000);
+ return 8000;
if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
payload_type == FLAG_avt_16)
- return rtc::Optional<int>(16000);
+ return 16000;
if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
- return rtc::Optional<int>(32000);
+ return 32000;
if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
- return rtc::Optional<int>(48000);
+ return 48000;
if (payload_type == FLAG_red)
- return rtc::Optional<int>(0);
- return rtc::Optional<int>();
+ return 0;
+ return rtc::nullopt;
}
// Class to let through only the packets with a given SSRC. Should be used as an
@@ -294,7 +294,7 @@
<< static_cast<int>(packet.header.payloadType) << ")"
<< std::endl;
}
- last_ssrc_ = rtc::Optional<uint32_t>(packet.header.ssrc);
+ last_ssrc_ = packet.header.ssrc;
if (other_callback_) {
other_callback_->AfterInsertPacket(packet, neteq);
}
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 363eb6b..dbf5ad4 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -816,7 +816,7 @@
if (!myCodec) {
CodecInst ci;
AudioCodingModule::Codec(_codecCntrA, &ci);
- myCodec = rtc::Optional<CodecInst>(ci);
+ myCodec = ci;
}
if (!_randomTest) {
diff --git a/modules/audio_coding/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc
index 73c8542..e9e9430 100644
--- a/modules/audio_coding/test/PCMFile.cc
+++ b/modules/audio_coding/test/PCMFile.cc
@@ -225,7 +225,7 @@
}
void PCMFile::SetNum10MsBlocksToRead(int value) {
- num_10ms_blocks_to_read_ = rtc::Optional<int>(value);
+ num_10ms_blocks_to_read_ = value;
}
} // namespace webrtc